Hey Dan,
That worked ... its stuttering .. but keeps playing .... thanks a ton !
osssink: wav file play fine
mp3 files, http links stutter but output sound when
buffer-time=1000 latency-time=100 sync=false
Is there any such fix for esdsink ??
Are there more properties that can be modified to stop the stuttering ??
regards,
raj
On Fri, Aug 1, 2008 at 12:52 PM, Daniel Charles <dcharlesm at gmail.com> wrote:
> Hi Raj,
>
> Have you tried sync=false in the osssink? I'm not sure that this is
> within your purposes but in some cases that removes the glitches.
>
> Daniel.
>
> On Fri, Aug 1, 2008 at 12:41 PM, Raj Swaminathan <rajshyam at gmail.com>
> wrote:
> >
> > Hi everyone,
> >
> > Im having stuttering and stopping issues with gstreamer on the OMAP
> 2430...
> > I am using an NFS mounted file system via ethernet ...
> >
> > osssink: WAV files play without an issue.
> > mp3 files output sound and stop after a few seconds if i
> set
> > buffer-time=1000 and latency-time=100
> > mp3 files do not output sound without the settings above.
> > streaming music from http links do not work under any
> > setting.
> >
> > esdsink: WAV files do not play.
> > mp3 files play nicely.
> > streaming music from http links output sound and stop after
> a
> > few seconds.
> >
> > My sources: filesrc, souphttpsrc
> > My decoders: wavparse, mad
> >
> > Ive experimented with placing queues before decoding and before sending
> > audio to the sink. Trying both ways or either/or, do not impact the
> output
> > signifcantly.
> > Can anyone provide some suggestions?
> >
> > Thanks for your help so far.
> >
> > regards,
> > raj
> >
> > On Thu, Jul 31, 2008 at 4:42 AM, Jan Schmidt <thaytan at noraisin.net>
> wrote:
> >>
> >> On Wed, 2008-07-30 at 08:50 -0700, Dennis Fleming wrote:
> >> > First off: Thank to you and Zhoa-Lang for getting back so quickly.
> >> > I'm so busy I forgot my manners.
> >> >
> >> > Testing to find the parameters I have I used decodebin, but in the
> >> > program itself uses playbin with the same effect. The only variation
> >> > is that I set the sink property to alsasink since that seems the only
> >> > way to set buffer-time and latency-time properties. Also, it seems
> >> > counter-intuitive to me that an uncompressed WAV file should have
> >> > problems keeping up while MP3s with the same sampling frequency and
> >> > word size have none. And yet the artifacts are indicative of dropped
> >> > buffers.
> >>
> >> If the bottleneck is retrieving data from the input location, then it's
> >> entirely feasible. What's your data store? SD card, NFS? A WAV file
> >> might be 10 or more times more data to read and cause read stalls, where
> >> the smaller mp3 can be read in fine and decoded in memory with no
> >> further problems.
> >>
> >> J.
> >>
> >> > Dennis
> >> >
> >> >
> >> > ----- Original Message ----
> >> > From: Thijs Vermeir <thijsvermeir at gmail.com>
> >> > To: Dennis Fleming <arsantiqua at sbcglobal.net>
> >> > Cc: gstreamer-embedded at lists.sourceforge.net
> >> > Sent: Tuesday, July 29, 2008 3:59:30 PM
> >> > Subject: Re: noise and stuttering
> >> >
> >> > Hi,
> >> >
> >> > On Tue, Jul 29, 2008 at 5:43 PM, Dennis Fleming
> >> > <arsantiqua at sbcglobal.net> wrote:
> >> > > The interesting thing is that uncompressed WAV files are causing the
> >> > problem
> >> > > while MP3s were fixed by setting the buffer-time and latency-time to
> >> > values
> >> > > smaller than found on a desktop. What would adding a queue do to
> >> > latency
> >> > > through the system?
> >> >
> >> > There is no latency in this case because there are no live-sources.
> >> > [1]
> >> >
> >> > > Also, I suppose, that I will need to break up the
> >> > > playbin and create a pipeline myself, yes?
> >> >
> >> > playbin has the queue elements on the correct location, no changes
> >> > needed.
> >> > You where already using a custom pipeline, no?
> >> >
> >> > Gr,
> >> >
> >> > [1]
> >> >
> >> >
> http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/draft-latency.txt?view=markup
> >> >
> >> > >
> >> > > Dennis
> >> > >
> >> > > ----- Original Message ----
> >> > > From: Thijs Vermeir <thijsvermeir at gmail.com>
> >> > > To: Zhao Liang-E3423C <E3423C at motorola.com>
> >> > > Cc: Dennis Fleming <arsantiqua at sbcglobal.net>;
> >> > > gstreamer-embedded at lists.sourceforge.net
> >> > > Sent: Tuesday, July 29, 2008 2:46:42 AM
> >> > > Subject: Re: noise and stuttering
> >> > >
> >> > > Hi,
> >> > >
> >> > > On Tue, Jul 29, 2008 at 11:15 AM, Zhao Liang-E3423C
> >> > <E3423C at motorola.com>
> >> > > wrote:
> >> > >> What's the rootcause of noise and stuttering ?
> >> > >
> >> > > Now you are using only 1 thread for all the elements and if the
> >> > > filesrc or the decoder is too slow sometimes
> >> > > you don't have time to catch up. By adding the queue you put the
> >> > sink
> >> > > in another thread and now the filesrc+decoder can
> >> > > do some decoding in advance.
> >> > >
> >> > > Gr,
> >> > > Thijs
> >> > >
> >> > >>
> >> > >> For normal playback, it should not have issues. If decoder didn't
> >> > drop
> >> > >> data, I think alsasink did it.
> >> > >> By gstaudiosink mechanism, it will drop data replaced with blank
> >> > data when
> >> > >> data is late. I guess the rootcause is that.
> >> > >>
> >> > >> If that, I have no ideas except adding a queue before alsasink, and
> >> > when
> >> > >> queue is empty, pause the pipeline, it will not cause dropout, but
> >> > still
> >> > >> discontinous.
> >> > >>
> >> > >> Zhao liang
> >> > >> ________________________________
> >> > >> From: gstreamer-embedded-bounces at lists.sourceforge.net
> >> > >> [mailto:gstreamer-embedded-bounces at lists.sourceforge.net] On
> Behalf
> >> > Of
> >> > >> Dennis Fleming
> >> > >> Sent: Tuesday, July 29, 2008 4:37 AM
> >> > >> To: gstreamer-embedded at lists.sourceforge.net
> >> > >> Subject: noise and stuttering
> >> > >>
> >> > >> I'm trying to create an audio player on an IMX31 target and I've
> >> > found a
> >> > >> discrepancy in the output of various formats. If I send MP3 data I
> >> > have
> >> > >> to
> >> > >> set the buffer-time and latency-time to 10000 and 100 respectively
> >> > to play
> >> > >> without severe dropouts. However WAV files still have drop-out at
> >> > a
> >> > >> consistent rate (about 1 per 10 sec). Are there some general
> >> > features I'm
> >> > >> missing or is there some guidance on the buffer-time/latency time
> >> > that
> >> > >> would
> >> > >> account for this difference?
> >> > >>
> >> > >> Linux 2.6.22.19
> >> > >> gstreamer 0.10.17 (open-embedded)
> >> > >> gst-launch filesrc location=<file> ! decodebin ! alsasink
> >> > >> buffer-time=10000
> >> > >> latency-time=100
> >> > >>
> >> > >> Dennis
> >> > >>
> >> > >>
> >> >
> >> >
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> >> >
> >> >
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> >> --
> >> Jan Schmidt <thaytan at noraisin.net>
> >>
> >>
> >>
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